March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to Manage a 2 Line Call
by Zoltan Medveczky 20 Mar '13

20 Mar '13

I'm trying to figure out the best way to implement the following scenario: 1) A Caller dials into FreeSWITCH and is bridged to an Agent phone. 2) At some point, the Agent puts the Caller on hold (with music) and dials out to his Supervisor. 3) After conversing with his Supervisor for some time, the Agent puts the Supervisor call leg on hold and takes the Caller off hold. 4) At this point, the Agent can put either party on hold while conversing with the other. Note that I don't need this to become a 3 way call at any time. Also, I'm controlling this scenario from an external application via "mod_event_socket" as opposed to using the XML dial plan. My question is, can this scenario be implemented simply using the "hold", "unhold", and "bridge" dial plan applications, or does this require the use of the conferencing module? I had some issues using the first approach, specifically, I was not able to initiate a dial out to the Supervisor phone while the Caller leg was on hold. Also, the Supervisor call leg would get hung up on whenever I tried to bridge back to the Caller. In regards to using the conferencing module, is it reasonably straight forward to move 2 call legs (i.e. the Caller and the Agent) from a bridged call into a conference? I could avoid bridging altogether and immediately conference the Caller and the Agent at the time that the Caller dials into FS, but I want to avoid the overhead involved in setting up a conference in the case that this 2 line call scenario is not required which is typically the case. Thanks, - Zoltan

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For calls within a single switch, is there any way to pass back name information of the called number?
by Nathan Neulinger 20 Mar '13

20 Mar '13

Or is this functionality solely of the phone itself - i.e. the directory on a polycom phone. On CCM, if you dial a local extension, the phone display updates to show the name/information for the called/destination number. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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How to properly use the "skinny-wait" target
by Nathan Neulinger 20 Mar '13

20 Mar '13

Looking at being able to handle a variable length dial pattern with a cisco phone (for international numbers). With a SIP phone, it's easy since the digits don't get sent incrementally, but with the SCCP phones, it seems like I get the digits one at a time, even if I dial them all and then lift the handset or press new call. This makes it so that a longer dial pattern will never match if a shorter one does. If I change the 'continue' in the skinny-wait action to false, or remove it - it never stops waiting for another digit. Worst case scenario - I could eliminate support for variable length dialing, and require a "#" at the end for international calls from the skinny phones, but is there a better way to accomplish this? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Hosted Freeswitch and Cisco ASA 5505
by Chris Ferreira 20 Mar '13

20 Mar '13

Hi Folks, I'm trying to preserve my sanity here and want to double check this. Are there any special considerations or configurations that need to be made when using a Cisco ASA 5505? A Cisco Tech is telling me everything is setup correctly and that no special NAT or Firewall rules need to be made. I say that's incorrect and have given them the info at http://wiki.freeswitch.org/wiki/Firewall on ports and protocols, etc. I swapped out the Cisco with a basic Linksys just to test and everything is fine. Thoughts? Thanks, -Chris

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How to play beep
by Muhammad Shahzad 20 Mar '13

20 Mar '13

Hi, How can i play beep e.g. before recording a file, i want to alert the user with beep. I thought its a sound file that installs with FS (like the one comes with Asterisk installation), but i couldn't find any such file in any format under default sounds. However, i do checked and confirmed that FS plays a beep when e.g. recording voicemail or user greeting etc. etc. but it does print file name it play for beep in console logs. What am i missing here? Thank you. -- Mit freundlichen Grüßen Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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FreeSWITCH Weekly Conference call for March 20th
by Ken Rice 20 Mar '13

20 Mar '13

Hey Guys, Join us Wednesday March 20th, 2013 at 1700 UTC (1200 CST / 1300 EST) for the weekly FreeSWITCH Community Conference Call. Access info and Agenda below. http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Legal Issues and others in Connecting a fixed line to Freeswitch in India
by Amit Kumar 20 Mar '13

20 Mar '13

What are the legal issues involved if I want to route all incoming calls to my fixed line number to an IVR, and to allow a program to make automated calls using the same fixed line? Also, can I use a PAP2 for the same? or do I need something like an SPA3102?

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FreeSwitch + A2Billing + mod_nibblebill
by Robert Huddleston 20 Mar '13

20 Mar '13

Anyone on the list have a wiki / primer / etc on integrating A2Billing with FreeSwitch? I got a good portion of it working with mod_xml_curl and mod_xml_cdr - however I'm left with also needing to add in mod_nibblebill for real-time cutoff / suspension of calls when the balance is depleted. It appears A2Billing's model (possible ASTPP as well) does not offer true real-time cutoff / suspension when multiple calls (concurrent / simultaneous) are in progress. Thanks

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mod_avmd help
by imperium broadcast 20 Mar '13

20 Mar '13

Hi, I have been testing mod_avmd, and out of the 200+ calls Ive managed to get it to work once, I have called two mobile networks and Our internal Freepbx with out much success. The set up for making an outbound call is as follow. Mobile Network: Freeswitch -> Freepbx -> Asterisk(DAHDI). Office PBX: Freeswitch -> Freepbx Here is my dial plan. <extension name="ast_extens"> <condition field="destination_number" expression="^(\d{11})$"> <action application="info"/> <action application="set" data="execute_on_answer=avmd"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1(a)10.2.1.140"/> <action application="info"/>  <action application="avmd" data="stop"/>  <action application="hangup"/> </condition> </extension> This is the output on the console http://pastebin.freeswitch.org/20705 Am I missing something any where ? Regards Impy

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ashish gautam has invited you to open a Google mail account
by ashish gautam 20 Mar '13

20 Mar '13

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